I am using a Hack RF One as SDR and the software GNU Radio to receive and process signals emitted from a drone. Given that the drone is moving and is emitting signals from a wide zone, received signal ...

I have to modell the quantization noise of ADC.
I only know that the maximum relative error is 1 dB.
Since this is logarithmic, I find it hard to formulate an appropriate model for the quantization ...

I have 2 oversampling ADC's running parallelly, each to process data in a specific range of the input as shown below:
Each ADC can process only half cycle range of a sine wave. Each ADC adds its own ...

Oversampling a signal means sampling it with a significantly higher sampling frequency than the Nyquist rate. As far as I know, there are three advantages:
Easier design of anti alias filter
Increase ...

I have optical OFDM baseband TX and RX implemented in FPGA. The design is tested using one FPGA board in loopback connection from DAC to ADC via coaxial cable. It works fine, the BER is about 1.10^-8 ...

I am using a Sigma Delta ADC and really understood the working of it
I need to sample a signal at 8.5 Mhz.
The SD ADC has a Demodulator and decimating FIR filter
The Oversampling ratio options are 32/...

I've RF data that I can see my packets inside by Matlab, the file of RF data is a raw data(IQ SAMPLES DATA) ....
I have did demodulation to those data which the output is also a demodulated raw data ...

i have a flash ADC converter with 1V vref and idial ressistors but in the comparator i have standart deviation of 3mV.
how do i calculate from the the standart deviation of the whole device(DNL INL)?
...

Can any one help me understand why there is a shift in Q values? It seems there is a DC offset/bias. How do I remove it or set it accordingly to make it equal to the I values/waveform? I am using a ...

I am trying to simulate an M-PSK Tx-Rx System on Simulink and analyse what the effect of sampling rate reduction would have on it. More, particular I am trying to prove what Nyquist sampling theorem ...

I have a design in which a signal from ADC is filtered and then played by DAC. In FPGA the real signal is converted to quadrature using DDS with 70 MHz frequency. After filtering, the baseband signal ...

Based on some reading, here's my understanding of how sound recording works:
Microphone outputs a voltage based on sound pressure. The microphone voltage is amplified and the amplified voltage goes ...

I am generating a sinusoid from a signal generator set at a specific $\textrm{dBm}$ and inputting this into an ADC as a real input (not IQ). I take a 256 point FFT on the resulting samples (raw signed ...

I have written Matlab script for BPSK modulation. The way my code works is as follow:
series of 0 and 1----> NRZ data (continuous rectangular pulses with amplitude -1 and 1)--->multiply by carrier----...

I read that in A/D converter we usually use rounding or two's complement truncation and not the sign magnitude or 1's complement truncation as in this case the quantization error is correlated to the ...

I have IQ samples data file, RF data, that I sampled it by dongle SDR RTL which this file has packets that I transmitted them and the dongle has sampled them, so the IQ samples data file which is the ...